Simultaneous recording and uploading of multiple audio files of the same conversation and audio drift normalization systems and methods

ABSTRACT

The invention relates to audio drift normalization, and more particularly to audio drift normalization systems and methods that can normalize audio drift of a plurality of recordings from a source.

FIELD OF THE INVENTION

The invention relates to simultaneous recording and uploading systems and methods, and, more particularly to a simultaneous recording and uploading of multiple files from the same conversation. Further, the present invention relates to audio drift normalization, and more particularly to audio drift normalization systems and methods that can normalize audio drift of a plurality of recordings from a source.

COPYRIGHT NOTICE

A portion of the disclosure of this patent document contains material that is subject to copyright protection. The copyright owner has no objection to the facsimile reproduction by anyone of the patent document or the patent disclosure, as it appears in the Patent and Trademark Office files or records, but otherwise reserves all copyrights whatsoever.

CROSS-REFERENCE TO RELATED APPLICATION

This application claims priority from U.S. Provisional Patent Application No. 62/947,490, filed Dec. 12, 2019, titled SIMULTANEOUS RECORDING AND UPLOADING OF MULTIPLE AUDIO FILES OF THE SAME CONVERSATION AND AUDIO DRIFT NORMALIZATION SYSTEMS AND METHODS and U.S. patent application Ser. No. 17/119,784, filed Dec. 11, 2020, the disclosure of which is incorporated by reference herein for all purposes.

BACKGROUND OF THE INVENTION

As advances in internet communications including video and audio mediums continue to increase, the need to effectively and efficiently communicate and create media across a broad spectrum of platforms has also increased. Many of the communications platforms configured to communicate recordings cannot ensure that the data is accessible in the event of a problem with the recording process and minimize or eliminate the need to record the complete session prior to uploading or produce quality media recordings.

Previous attempts to create systems and methods to ensure that the data is accessible in the event of a problem with the recording process and minimize or eliminate the need to record the complete session prior to uploading or produce quality media recordings, have been unsuccessful, time consuming and costly.

These previous attempts have necessitated additional hardware, systems, time expenditure, and costs. Previous attempts to address these problems have not been efficient or successful.

In many instances, podcasters, audio book producers, and content creators find themselves spending time, effort, and extra costs to produce quality video and/or audio recordings. These additional logistical steps increase operating costs and increase the time needed to publish video and/or audio works. These previous methods needed to record an audio and/or video file in its entirety before being uploaded to the cloud.

Further, these methods require additional system resources to be present within a recording device thereby increasing overhead of systems and methods.

Quite often recording professionals find themselves spending additional time waiting for large files to upload after a recording is finished.

Many of the communications platforms configured to communicate recordings require systems, methods, and logistical intervention to correct or adjust recordings for irregularities. Some of these irregularities include drift between multiple collaborators, lack of synchronization, and data inconsistency.

Previous attempts to create systems and methods to correct or minimize drift in recordings have necessitated additional hardware, systems, time expenditure, and costs. Previous attempts to address these problems have not been successful.

Further, imperfections in recordings can cause drift. Specifically, audio drift requires efforts to realign each audio file to the actual timing for it to be usable in products, adding time and work for producers and audio engineers in post-production.

Quite often recording professionals find themselves realigning audio recordings in post-production to correct recordings with drift or that are out of sync.

When multiple systems are recording, and two or more audio recordings of the same conversation are captured on multiple devices, reconstruction and integration of the recordings require a process to normalize the drift that arises between the two or more audio recordings. Therefore, audio drift results in the need to realign each audio file to the actual timing for a usable product. This adds time and work for producers and audio engineers in post-production.

Another problem can occur when the recording process utilizes segments or chunks of data based on a preset time period and the recording being transferred at different times. The recording assembly system must take all the chunks and write them into one file. In doing so they need to ensure that the data is captured, is continuous, and mixed together. This assembly can result in audio drift.

Additionally, combining recordings from a plurality of sources often requires alignment of the waveforms to eliminate the over writing of each other when they are out of phase. This condition requires additional steps, costs, and time to publish works.

What is needed in the art is a solution to address issues with recording and uploading of audio and video files. Additionally, there is a need for a solution to address audio drift issues.

SUMMARY OF THE INVENTION

The invention in one form is directed to a method that ensures that recorded audio and video files are always accessible in the event of a catastrophe, removes the need for the audio and video file to exist in its entirety before uploading and requires less system resources be present within the recording device.

The invention in another form improves the reliable access to complete audio files and reduces time waiting for large files to upload after a recording is finished.

The invention in yet another form allows for the use of low-end devices due to the efficiency of the recording system.

In yet another form, the invention improves over existing chunk style recording methods.

In another embodiment, the system can also include algorithms and/or software which eliminate manual processes to perform system processes and utilize methods.

In yet another embodiment, the system can include methods to perform chunk style recording to produce synchronized recordings.

In an aspect, the system can include processes to normalize audio drift that arises between two or more audio recordings of an original source that is captured on one or a plurality of devices.

In another aspect, the system can also include components that are configured to minimize human interaction and/or executed steps to produce enhanced audio and video quality recordings. These recordings are designed to produce top grade audio recordings, the recordings configured to be combined in other audio and/or video recordings without the need for back end engineering to improve quality of the recordings.

In yet another aspect, the system can include systems configured to provide near real-time interview capability with video and record everyone's audio in separate media files without drift.

In an embodiment, the system can include a platform which provides a medium to connect podcast hosts, co-hosts, & guests to record studio quality audio and video from anywhere.

In another embodiment, the system can also include allowing system users to download, listen, and share one or all of user's recordings organized by any parameter, including but not limited to session, date, geography and/or recording medium.

In yet another embodiment, the system can also include a video feature so a podcaster and a guest can see each other while recording.

In an aspect, the system can include mobile applications and can be utilized on a mobile device.

In another aspect, the system can also include laptops, desktop computers, and computing mediums.

In yet another aspect, the system can include utilizing a standardized 48 kHz sampling rate.

In an embodiment, the system can include processes which do not require echo cancellation (Acoustic Echo Cancellation (AEC) Software).

In another embodiment, the system can also include remote audio signals and can exist on the cloud/web within a network.

In yet another embodiment, the system can include saving recordings in buffers of any length of time, preferably between 2 and 8 seconds. The system can also utilize resampling to 48 kHz.

In an aspect, the system can include longform recordings, exceeding one hour in length.

In an aspect, the system can reduce the need for local storage and is unbounded in the duration of recording.

In another aspect, the system can also include components and/or methods which are non-destructive wherein no data is lost.

In yet another aspect, the system can include systems and/or methods that do not utilize a master clock or drift threshold.

In an embodiment, the system can include methods and/or components which normalize audio drift from different sources.

In another embodiment, the system can include quality, reliable, repeatable, systems efficient enough to utilize low end devices.

In another embodiment, the system can be used alongside aviation flight data recording systems to reduce the need to search for the “black box” because the recordings are not stored locally.

According to an aspect of the present invention, an audio drift normalization system is presented. The system can ensure that recorded audio files are automatically modified to correct audio drift arising from a plurality of recording devices and/or buffer size differences.

In another embodiment, the system can also include algorithms and/or software which eliminate manual processes to correct audio drift.

In yet another embodiment, the system can include methods to perform chunk style recording to produce synchronized recordings.

In an aspect, the system can include processes to normalize audio drift that arises between two or more audio recordings of an original source that is captured on one or a plurality of devices.

These and other features and advantages will become apparent from the following detailed description of illustrative embodiments thereof, which is to be read in connection with the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

The features and advantages of the present invention will be better understood when the Detailed Description of the Preferred Embodiments given below is considered in conjunction with the figures provided.

FIG. 1 is a method flowchart for simultaneous recording audio files in accordance with an embodiment of the present invention;

FIG. 2 is a method flowchart for simultaneous recording audio files in accordance with an embodiment of the present invention; and

FIG. 3 is a method flow chart for audio drift normalization in accordance with an embodiment of the present invention.

FIG. 4 is method flow chart for progressive rendering of audio and video chunks.

DETAILED DESCRIPTION

The present invention will now be described more fully hereinafter with reference to the accompanying drawings, in which embodiments of the invention are shown. This invention may, however, be embodied in many different forms and should not be construed as limited to the embodiments set forth herein. Rather, these embodiments are provided so that this disclosure will be thorough and complete, and will fully convey the scope of the invention to those skilled in the art. Like numbers refer to like elements throughout.

With respect to the term chunks when used by the instant invention may refer to segments of audio and/or video recordings.

The terms media recordings, audio recordings, video recordings may be used interchangeably and mean a recording of the using a media capture device such as a microphone or video camera.

The terms media data media files, audio data, audio files, video data, video files may be used interchangeably and mean data stored on a computer representing the data of the captured media.

The term WAV, MP3, MPS, audio, video file, AVI (Audio Video Interleave), FLV (Flash Video Format), WMV (Windows Media Video), MOV (Apple QuickTime Movie), MP4 (Moving Pictures Expert Group 4) may be used interchangeably within the specification and are all forms of media files.

The term cloud when used in context by the instant invention can mean Amazon cloud, Google cloud, Verizon cloud or any suitable cloud storage system.

The term computing device capable may include any type of device for computing, such as a tablet, mobile device, smart speaker, flight data recorder, or computer that may be in communication with a media recording device such as a microphone or video camera.

The term public URL is a URL that can be asked by anyone on the Internet as it is indexed by search engines.

The term video file format is a type of file format for storing digital video data on a storage device such as a computer system. Video is almost always stored using compression to reduce the file size. A video file normally consists of a container containing video data in a video coding format alongside audio data in an audio coding format.

The term stereo commonly refers to Stereophonic sound and is the reproduction of sound, using two or more independent audio channels, through a symmetrical configuration of loudspeakers, in such a way as to create a pleasant and natural impression of sound heard from various directions, as in natural hearing.

The term monophonic sound commonly called mono sound, mono, or non-stereo sound; this early sound system used a single channel of audio for sound output. In monophonic sound systems, the signal sent to the sound system encodes one single stream of sound and it usually uses just one speaker or the same exact sound in both speakers.

When the term database is used within the specification it can alternatively mean any suitable storage medium can be substituted such as a data structures, files, stacks utilizing Direct access storage device (DASD) or non-volatile computer storage an example of which is solid state storage medium.

The term audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data (excluding metadata) is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer.

Typical audio recording devices tend to sample audio at one of two different rates, 44.1 kHz or 48 kHz.

There are three major groups of audio file formats:

Uncompressed audio formats, such as WAV, AIFF, AU or raw header-less PCM;

Formats with lossless compression, such as FLAC, Monkey's Audio (filename extension .ape), WavPack (filename extension .wv), TTA, ATRAC Advanced Lossless, ALAC (filename extension .m4a), MPEG-4 SLS, MPEG-4 ALS, MPEG-4 DST, Windows Media Audio Lossless (WMA Lossless), and Shorten (SHN).

Formats with lossy compression, such as Opus, MP3, Vorbis, Musepack, AAC, ATRAC and Windows Media Audio Lossy (WMA lossy).

Specifically, Opus is a lossy audio coding format and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. However, when used in the specification the invention can utilize any lossy compression method.

Any embodiment of the invention disclosed can be used with any media, audio or video file format and all terms are used interchangeably.

Embodiments of the invention can be divided into three separate applications. The first is within a Client Server software architecture, i.e., a web application, and the second is within a cloud-based Server which sends and or receives data from the client server software, and the third can be the audio drift normalization where the audio is synchronized. However, the audio drift normalization can be implemented with a different architecture which can provide required inputs and is capable of utilizing the specific output of the audio drift normalization application.

Typical audio drift algorithms can be selected from the group consisting of drift correction modulation, clock drift compensation techniques and multimedia decoding device. European patent specification EP 2038886 B1 inventor Abrol publication date Nov. 4, 2012 discloses various techniques to achieve clock drift compensation for audio decoding and the disclosure of which is incorporated by reference herein for all purposes.

Referring now to the Client Server software architecture there is shown a Web Application Client running in a Web Browser.

A) Permission is requested for access to a Microphone and/or Video Camera Media Device

B) Upon permission being granted, a Media Stream (https://w3c.github.io/mediacapture-main/#stream-api) is initialized

C) The Media Stream is provided to a Media Recorder (https://www.w3.org/TR/mediastream-recording)

D) Upon Start recording:

-   -   a. the Media Recorder is configured to slice “Chunks” of the         recording into Chunks of data which can be saved in any         reasonable time length of the recording which is preferably from         0.5 to 60 seconds and is not to exceed 1 Megabyte (MB) in size.         However, the most preferable length is 8 seconds,     -   b. the File_name is created.

Every 8 seconds thereafter each “Chunk” is:

-   -   a. Encoded in Base64 format to convert it from a raw Array         Buffer to a String,     -   b. Given a Date Time Stamp,     -   c. Saved to a Collection in a cloud database such as but not         limited to Google Cloud Platform's Firestore Database.

Upon Stop recording an HTTP Post Request is sent to a server running in the Cloud containing:

-   -   a. The creator's identification/authorization token.     -   b. The reference path to the Collection on “Chunks” within a         database     -   c. The File_name.

A POST request is an arbitrary amount of data of any type that can be sent to the server in the body of the request message. A header field in the POST request usually indicates the message body's Internet media type.

Therefore, referring to the Server running in the Cloud:

-   -   a. The reference path to the Collection on “Chunks” within         Google Cloud Platform's Firestore Database is used to Get the         “Chunks” ordered by the Date Time Stamp.     -   b. Each “Chunk” is Decoded from Base64 back to Array Buffer.     -   c. The Array Buffers are concatenated into a File Buffer.     -   d. The File Buffer is saved to disk using a multi-media         container format with the provided File_name.     -   e. Audio Drift Normalization—the saved file from the multi-media         container can be converted to an audio and/or video file format         and then is normalized without reduction of quality using an         Audio drift normalization application.     -   f. Media files are uploaded to a Cloud Storage Bucket within         Google Cloud Platform with Public URLs being provided in return.         The media formats can alternatively be any audio or video file         format which provides an efficient data encoding format.     -   g. The Public URLs of the media files are saved to the         Application Data within a database such as Google Cloud         Platform's Firebase Real-Time Database using the Reference Path.

Referring to the audio drift normalization phase:

Audio recording devices tend to sample audio at one of two different rates, 44.1 kHz or 48 kHz. Audio files encoded with the Opus codec/encoder and stored within the WebM media file container format is processed by the FFMPEG open source audio/video utility in the following ways to standardize the output files.

-   -   a. If the Sample Rate is 44.1 kHz then resampled it to 48 kHz;     -   b. Synchronizing the wave forms if the input is stereo then sum         it to mono;     -   c. Converts to WAV and MP3 container formats;     -   d. Saves WAV and MP3 files to disk.

Clarifying binary-to-text encoding schemes, Base64 is designed to carry data stored in binary formats across channels that only reliably support text content. Base64 is particularly prevalent on the World Wide Web where its uses include the ability to embed image files or other binary assets inside textual assets such as Hypertext Markup Language (HTML) and Cascading Style Sheets (CSS) files. However, any similar application/product could be used to achieve similar results.

It should be noted that WebM is an open, royalty-free, media file format designed for the web. WebM defines the file container structure, video and audio formats. WebM files consist of video streams compressed with the VP8 or VP9 video codecs and audio streams compressed with the Vorbis audio compression or Opus audio coder-decoder which is a real-time interactive audio coder-decoder. However, any similar application/product could be used to achieve similar results.

A WAV file is a raw audio format that are uncompressed lossless audio and as such can take up quite a bit of space, coming in around 10 MB per minute with a maximum file size of 4 GB. However, any similar application/product could be used to achieve similar results.

An MP3 file is an audio file saved in a compressed audio format developed by the Moving Picture Experts Group (MPEG) that uses “Layer 3” audio compression. However, any similar application/product could be used to achieve similar results.

WAV and MP3 are protocols for converting the audio or video data to the associated file formats.

The term WAV, MPS, audio, video file, AVI (Audio Video Interleave), FLV (Flash Video Format), WMV (Windows Media Video), MOV (Apple QuickTime Movie), MP4 (Moving Pictures Expert Group 4) are used interchangeably within the specification and are all forms of media files.

The term codec is a device or computer program for encoding or decoding a digital data stream or signal. Codec is a portmanteau of coder-decoder. A coder encodes a data stream or a signal for transmission or storage, possibly in encrypted form, and the decoder function reverses the encoding for playback or editing.

The term FFMPEG is a free and open-source project consisting of a vast software suite of libraries and programs for handling video, audio, and other multimedia files and streams.

An alternate embodiment of the instant invention can be used to provide progressive rendering of audio and video chunks.

On frontend web application, send a HTTP POST request to the backend web service after 2 or more audio and/or video chunks are uploaded which equals 24 seconds after the Start Recording event is fired to begin rendering the initial audio and video chunks.

Using the backend web service, create a new queue in Cloud Tasks Manager for asynchronous execution for each file.

On the frontend web application, enqueue a new task for each file/chunk by sending an HTTP POST request from said frontend web application to backend web service.

On Google Cloud Tasks, deque tasks to append that file/chunk to a previously rendered audio and video files.

On backend web service

-   -   a. Download the previously rendered audio or video file     -   b. Save it to the/temp folder     -   c. Concatenate the previously rendered file with the chunk file     -   d. Save the concatenated rendered audio file and video file it         to the/temp folder or upload to a cloud storage     -   e. Complete the task using Google Cloud Tasks Manager

Repeat 3-5 until there are no more chunk rendering tasks are in the Cloud Tasks manager for asynchronous execution queue/Task Manager Queue.

An second alternate embodiment of the instant invention can be used to provide progressive rendering of audio and video chunks and a cloud application with a frontend web application and backend web service.

The second alternate embodiment method of progressive rendering of a recording event wherein the recording event comprises of at least one recorded media file having a recorded media file start recording event and a stop recoding event and the at least one recorded media file is recorded using a first recording device using a cloud application and the cloud application having a frontend web application and a backend web service, the method comprising:

-   -   step a, the frontend web application sending an HTTP POST         request to an backend web service after 2 or more audio and/or         video chunks are uploaded or approximately 24 seconds after the         at least one recorded media file the start recording event is         started and the cloud application using the backend web service         rendering the at least one recorded media file;     -   step b, the cloud application using the backend web service         creating a new queue in Cloud Tasks manager for asynchronous         execution for each file/chunk;     -   step c, the cloud application using the frontend web application         having a task queue and enqueueing a new task in the task queue         for each file/chunk by sending an HTTP POST request from the         frontend web application to backend web service;     -   step d, Using a Cloud Tasks Manager for asynchronous execution,         deque tasks to append the chunk to a previously rendered audio         and video file.     -   step e, the cloud application using the backend web service:     -   downloading the previously rendered audio or video file;     -   saving the chunk to a temporary folder on the cloud application;     -   concatenating the previously rendered audio and video file with         the chunk file thereby forming a concatenate rendered audio and         video file to cloud storage;     -   saving the concatenated rendered audio and video file with the         chunk file to the temporary folder on the cloud application;     -   uploading the concatenate rendered audio and video file with the         chunk file cloud storage;

Completing the task using Cloud Tasks manager for asynchronous execution;

Repeat steps c through step e until there are no more chunk rendering tasks in the task queue.

Referring now to the drawings as shown in FIGS. 1, 2, 3, and 4 and more particularly to FIG. 1 , there is shown a Client Server software architecture of the invention web application 1000. The server side is shown in FIG. 2 server application. By splitting the invention into two separate components, client side and web side, the invention increases the efficiency of the transfer.

The application of the invention starts at step 10 which resides on the client side of the application on a web application running on a computing device. The client computing device being in communication with a microphone. The client application then requests permission to access the microphone step 20. Once permission has been received to access the microphone or video capture device, the media stream is initialized by the web application 1000 as shown in step 30. The media recorder 40 is enabled so as to be capable of receiving the microphone and/or video camera media stream. The microphone and/or video camera sends a start recoding signal. When the start recording signal step 50 is received by the web application 1000, then the client side web application opens a file step 60 and starts media recorder that records data from the microphone and/or camera media stream and stores it in the file. The web application 1000 then creates a chunk and appends a name to the file for identification purposes. The segments are based on a preset time period. The recorded data is then segmented into chunks of data step 70. Chunks of data can be saved in any reasonable time length of the recording which is preferably from 0.5 to 60 seconds. However, the most preferable length is 8 seconds and is not to exceed 1 Megabyte (MB) in size.

The recorded data is recorded until the recording device sends stop recording signal to receive the web application 1000. Then the chunks of data represented by the files with the name created in step 60 and an appended sequence number which starts a 1 and the sequence number stop with an integer value equal to the number of files, therefore sequence number two would equal 1+1 which equals 2. The files are then converted to Base64 encoded raw data chunk and placed in an array buffer in step 80 using the name assigned to the chunk in step 70. The Base64 encoded raw data chunk is then encoded with a date and time stamp in step 90 which is the transfer chunk and that transfer chunk is uploaded to the cloud and stored in a Firestore database. However, any similar database application/product could be used to achieve similar results such as Oracle, Zoho Creator, Quick Base, Amazon Aurora, Kintone, Amazon RDS, Rubrik, Airtable, Knack, TeamDesk, Firebolt or Firestore.

The media recorder step 40 has a second feature which is continually sensing the device for the stop recording signal step 120. Once the stop recording signal is received, the web application 1000 processes and sends HTTP Post request step 110 to the cloud server where the chunks are stored in the Firestore database via link A.

Referring now to FIG. 2 , there is shown the cloud server application where the chunks of data are stored. Step 210 receives the HTTP post request sent from the client application step 110 on FIG. 1 and retrieves the data from the Firestore database step 220. The Firestore database can be replaced with any similar application capable of storing text, binary, ASCII, audio or video files. Once all the chunks of data have been received, the files are organized by the name created in step 60 of FIG. 1 in step 230. Step 240 the application 2000 decodes the Base64 and stores the results in an array buffer. The application 2000 does this for all the chunks of step 230 until the complete recording is decoded and each file is identified by the name created in step 60 of FIG. 1 in step 230 plus a sequence number or identifier. Step 250 takes decoded data from step 240 and concatenates the files into a file buffer where the media, audio or video file is recreated. Once the concatenated file is created it is stored to the server storage medium step 260 in an audiovisual media file format such as webm. The server storage medium can be any suitable technology including random and serial data storage media such as a disk drive or electronic media accessible by the server running application 2000. The storage medium does not have to be contained in the same server but can be a networked device. In step 270 the file stored in an audiovisual media file format in step 260 which is identified by the name created in step 60 of FIG. 1 plus a sequence number or identifier assigned in step 230. Step 280 of application 2000 then operates on the file from step 270 and creates a WAV and or MP3 file in step 290. The term WAV, MPS, audio, video file, AVI (Audio Video Interleave), FLV (Flash Video Format), WMV (Windows Media Video), MOV (Apple QuickTime Movie), MP4 (Moving Pictures Expert Group 4) are used interchangeably within the specification and are all forms of media files. Step 300 then stores the newly created WAV or MP3 file in a cloud storage bucket. The term cloud storage buckets are the basic containers on the Google or Amazon cloud that contains the data. However, any suitable storage medium could be used. Step 310 then stores the public URL of the cloud storage bucket to the Firebase real time database however any suitable database application can be employed. Once the WAV or MP3 file and associated public URL are stored the application 2000 ends step 320.

Referring to FIG. 3 , there is shown the audio drift normalization process 280 from FIG. 2 , The process begins in step 400/here the process receives from step 270 of application 2000 the audiovisual media file format stored in step 260 as shown in FIG. 2 . The audiovisual media file format can be any file format which is applicable such as WebM, AVI, MPEG transport stream, Matroska, F4V, Vob, Ogg Video, 3GPP2, SVI, AMV video format, MPEG-1, MPEG-4, MPEG transport stream, QuickTime file format and Windows media video. The process then places the file into queue 410 within the server storage system comprising of a direct access storage device (DASD) or non-volatile computer storage, an example of which is solid state storage medium for further processing. Step 420 utilizes the WebM container media file format files and then processes them with the FFMPEG open source audio and video utility in the following ways to standardize the output files. In step 430 the process determines the sample rate of the file. If the sample rate is not 48 kHz then the file is resampled to 48 kHz in step 440. Resampling, in terms of audio files, is also known as Sample Rate Conversion. This is done when you need to convert a digital audio file from a given sample rate into a different sample rate. Sample rate is the number of samples of audio carried per second, measured in Hz or kHz (one kHz being 1,000 Hz). If the file is 48 kHz, the file is then passed to step 450. The audio drift normalization process has now synchronized the wave forms and determines if the input file is in stereo or mono 450. If the input file is stereo, the process sums the file to mono 460 else the file is passed to step 470. Summing a stereo media file to mono means taking both channels of your stereo media file (left and right) and creating a composite single mono media file. In step 470, the file is converted to media file such as a WAV and MP3 container formats. However, any applicable media file such as a WAV, MPS, AVI, FLV, WMV, MOV, MP3 and MP4 is acceptable data. The process then saves the media file to disk step 480. However, any suitable storage medium can be substituted such as a database, files or stacks all of which can reside on a Direct Access Storage Device (DASD) or non-volatile computer storage such as a solid state storage medium. The WAV and MP3 files are stored in separate files step 490 for further processing. The process then returns control of the file to application 2000 step 280 as shown in FIG. 2 .

The audio drift normalization can be implemented separately from the recording applications 1000 and 2000 shown in FIG. 1 and FIG. 2 . The utilizing application must provide the correct file format and be able to receive the file format being returned from application 3000.

Referring to FIG. 4 is a method flow chart for progressive rendering of audio and video. Step 510, is the start of the process. Step 520, The frontend web application, send a HTTP POST request to backend web service after 2 or more audio and/or video chunks are uploaded which is approximately 24 seconds after the Start Recording event is fired. Step 530 begin rendering the initial audio and video chunks. Then Step 540, Using the backend web service, create a new queue in a Cloud Tasks Manager for asynchronous execution for each file/chunk. Step 550, on the frontend web application enqueue a new task for each file/chunk by sending an HTTP POST request from said frontend web application to backend web service. Then Step 560, On Cloud Tasks Manager for asynchronous execution, deque tasks to append that file/chunk to the previously rendered audio and video files.

Step 570, On the backend web service:

-   -   a. Download the previously rendered audio or video file     -   b. Save it to the/temp folder     -   c. Concatenate the previously rendered file with the chunk file     -   d. Save it to the temp folder on the cloud application     -   e. concatenating said previously rendered audio and video file         with the chunk file thereby forming a concatenate rendered audio         and video file with the chunk file;     -   f. saving said concatenated rendered audio and video file with         the chunk file to said temporary folder;     -   g. uploading said concatenated rendered audio and video file to         the cloud storage;     -   h. completing the task using Cloud Tasks Manager;

Repeat Steps 550 to 570 until there are no more chunk rendering tasks are in the Cloud Tasks Manager Queue.

Step 590 the process stops.

The method shown in FIG. 4 , wherein the at least one recorded media file comprises and audio chunk or video chunk.

The method shown in FIG. 4 , wherein the audio chunk is selected from the group consisting of a WAV, MPS, AVI, FLV, WMV, MOV, MP3 and MP4 formatted data.

The method shown in FIG. 4 , wherein the video chunk is selected from the group consisting of WebM, AVI, MPEG transport stream, Matroska, F4V, Vob, Ogg Video, 3GPP2, SVI, AMV video format, MPEG-1, MPEG-4, MPEG transport stream, QuickTime file format and Windows media video.

In some embodiments, the system, method or methods described above may be executed or carried out by a computing system including a tangible computer-readable storage medium, also described herein as a storage machine, that holds machine-readable instructions executable by a logic machine (i.e. a processor or programmable control device) to provide, implement, perform, and/or enact the above described methods, processes and/or tasks. When such methods and processes are implemented, the state of the storage machine may be changed to hold different data. For example, the storage machine may include memory devices such as various hard disk drives, CD, flash drives, cloud storage, or DVD devices. The logic machine may execute machine-readable instructions via one or more physical information and/or logic processing devices. For example, the logic machine may be configured to execute instructions to perform tasks for a computer program. The logic machine may include one or more processors to execute the machine-readable instructions. The computing system may include a display subsystem to display a graphical user interface (GUI) or any visual element of the methods or processes described above. For example, the display subsystem, storage machine, and logic machine may be integrated such that the above method may be executed while visual elements of the disclosed system and/or method are displayed on a display screen for user consumption. The computing system may include an input subsystem that receives user input. The input subsystem may be configured to connect to and receive input from devices such as a mouse, keyboard, gaming controller microphone or camera. For example, a user input may indicate a request that a certain task is to be executed by the computing system, such as requesting the computing system to display any of the above described information, or requesting that the user inputs updates or modifies existing stored information for processing. A communication subsystem may allow the methods described above to be executed or provided over a computer network. For example, the communication subsystem may be configured to enable the computing system to communicate with a plurality of personal computing devices. The communication subsystem may include wired and/or wireless communication devices to facilitate networked communication. The described methods or processes may be executed, provided, or implemented for a user or one or more computing devices via a computer-program product such as via an application programming interface (API).

Since many modifications, variations, and changes in detail can be made to the described embodiments of the invention, it is intended that all matters in the foregoing description and shown in the accompanying drawings be interpreted as illustrative and not in a limiting sense. Furthermore, it is understood that any of the features presented in the embodiments may be integrated into any of the other embodiments unless explicitly stated otherwise. The scope of the invention should be determined by the appended claims and their legal equivalents.

The present invention has been described with reference to embodiments, it should be noted and understood that various modifications and variations can be crafted by those skilled in the art without departing from the scope and spirit of the invention. Accordingly, the foregoing disclosure should be interpreted as illustrative only and is not to be interpreted in a limiting sense. Further it is intended that any other embodiments of the present invention that result from any changes in application or method of use or operation, method of manufacture, shape, size, or materials which are not specified within the detailed written description or illustrations contained herein are considered within the scope of the present invention.

Insofar as the description above and the accompanying drawings, disclose any additional subject matter that is not within the scope of the claims below, the inventions are not dedicated to the public and the right to file one or more applications to claim such additional inventions is reserved.

Although very narrow claims are presented herein, it should be recognized that the scope of this invention is much broader than presented by the claim. It is intended that broader claims will be submitted in an application that claims the benefit of priority from this application. 

What is claimed is:
 1. A method of progressive rendering of a recording event wherein said recording event comprises of at least one recorded media file having a recorded media file start recording event and a stop recording event and said at least one recorded media file is recorded using a first recording device using a cloud application and said cloud application having a frontend web application and a backend web service, the method comprising: step a, sending an HTTP POST request to said backend web service after 2 or more audio and/or video chunks of a plurality of audio and/or video chunks are uploaded after said at least one recorded media file said start recording event is started and said cloud application using said backend web service rendering said at least one recorded media file; step b, using said backend web service creating a new queue in a Cloud Tasks Manager for asynchronous execution for each audio and/or video chunk of the plurality of audio and/or video chunks; step c, using said frontend web application having a task queue and enqueueing a new task in said task queue for each said audio and/or video chunk by sending a second HTTP POST request from said frontend web application to said backend web service; step d, using said Cloud Tasks Manager for asynchronous execution, deque tasks to append each said audio and/or video chunk to a previously rendered audio or video file; step e, using said backend web service to perform the steps: i. downloading said previously rendered audio or video file; ii. saving each said audio and/or video chunk to a temporary folder on a cloud application disk; iii. concatenating said previously rendered audio or video file with each said audio and/or video chunk in said temporary folder thereby forming a concatenated rendered audio and/or video file with the chunk file; iv. saving said concatenated rendered audio and/or video file with the chunk file to said temporary folder on said cloud application disk; v. uploading said concatenated rendered audio and/or video file with the chunk file to cloud storage; vi. completing said new task using said Cloud Tasks Manager; Repeat step c through step e until there are no more chunk rendering tasks in said task queue.
 2. The method of claim 1 wherein said at least one recorded media file comprises audio chunk or video chunk.
 3. The method of claim 2, wherein said audio chunk is selected from the group consisting of a WAV, MPS, AVI, FLV, WMV, MOV, MP3, FLAC, WebM, and MP4 formatted data.
 4. The method of claim 2, wherein said video chunk is selected from the group consisting of WebM, AVI, MPEG transport stream, Matroska, F4V, Vob, Ogg Video, 3GPP2, SVI, AMV video format, MPEG-1, MPEG-4, MPEG transport stream, QuickTime file format and Windows media video. 